Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index f7cdd827a2f71b40d15150d079d5c7291413fcc0..eba0c96c319cf7a11a3c1e45ca87cfd9737f0aec 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -511,15 +511,7 @@ Call::Stats Call::GetStats() const {
   stats.send_bandwidth_bps = send_bandwidth;
   stats.recv_bandwidth_bps = recv_bandwidth;
   stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs();
-  {
-    ReadLockScoped read_lock(*send_crit_);
-    // TODO(solenberg): Add audio send streams.
-    for (const auto& kv : video_send_ssrcs_) {
-      int rtt_ms = kv.second->GetRtt();
-      if (rtt_ms > 0)
-        stats.rtt_ms = rtt_ms;
-    }
-  }
+  stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
   return stats;
 }